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Commit 69933253 authored by Alexandre Savard's avatar Alexandre Savard
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[#4233] Add two incoming call test

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sflphone-common/test/sippxml/test_2.xml 0 → 100644
+ 161
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View file @ 69933253
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Two simultaneous outgoing call Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- Here is where the call is put on hold -->
<recv request="INVITE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
sflphone-common/test/siptest.cpp
+ 7
− 4
View file @ 69933253
...@@ -254,13 +254,18 @@ void SIPTest::testTwoIncomingIpCall () ...@@ -254,13 +254,18 @@ void SIPTest::testTwoIncomingIpCall ()
pthread_t firstCallThread, secondCallThread; pthread_t firstCallThread, secondCallThread;
void *status; void *status;
pthread_attr_t attr;
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
// the first call is supposed to be put on hold when answering teh second incoming call // the first call is supposed to be put on hold when answering teh second incoming call
std::string firstCallCommand("sipp -sf sippxml/test_2.xml 127.0.0.1 -i 127.0.0.1 -p 5064 -m 1"); std::string firstCallCommand("sipp -sf sippxml/test_2.xml 127.0.0.1 -i 127.0.0.1 -p 5064 -m 1");
// command to be executed by the thread, user agent client which initiate a call and hangup // command to be executed by the thread, user agent client which initiate a call and hangup
std::string secondCallCommand("sipp -sn uac 127.0.0.1 -i 127.0.0.1 -p 5062 -m 1 -d 1000"); std::string secondCallCommand("sipp -sn uac 127.0.0.1 -i 127.0.0.1 -p 5062 -m 1 -d 1000");
int rc = pthread_create(&firstCallThread, NULL, sippThread, (void *)(&firstCallCommand)); int rc = pthread_create(&firstCallThread, &attr, sippThread, (void *)(&firstCallCommand));
if (rc) { if (rc) {
std::cout << "SIPTest: ERROR; return code from pthread_create()" << std::endl; std::cout << "SIPTest: ERROR; return code from pthread_create()" << std::endl;
} }
...@@ -282,7 +287,7 @@ void SIPTest::testTwoIncomingIpCall () ...@@ -282,7 +287,7 @@ void SIPTest::testTwoIncomingIpCall ()
sleep(1); sleep(1);
rc = pthread_create(&secondCallThread, NULL, sippThread, (void *)(&secondCallCommand)); rc = pthread_create(&secondCallThread, &attr, sippThread, (void *)(&secondCallCommand));
if(rc) { if(rc) {
std::cout << "SIPTest: Error; return code from pthread_create()" << std::endl; std::cout << "SIPTest: Error; return code from pthread_create()" << std::endl;
} }
...@@ -313,7 +318,5 @@ void SIPTest::testTwoIncomingIpCall () ...@@ -313,7 +318,5 @@ void SIPTest::testTwoIncomingIpCall ()
std::cout << "SIPTest: completed join with thread 2" << std::endl; std::cout << "SIPTest: completed join with thread 2" << std::endl;
std::cout << "---------------------------" << std::endl;
} }
sflphone-common/test/siptest.h
+ 4
− 4
View file @ 69933253
...@@ -53,10 +53,10 @@ class SIPTest : public CppUnit::TestCase { ...@@ -53,10 +53,10 @@ class SIPTest : public CppUnit::TestCase {
* Use cppunit library macros to add unit test the factory * Use cppunit library macros to add unit test the factory
*/ */
CPPUNIT_TEST_SUITE( SIPTest ); CPPUNIT_TEST_SUITE( SIPTest );
// CPPUNIT_TEST ( testSimpleOutgoingIpCall ); CPPUNIT_TEST ( testSimpleOutgoingIpCall );
// CPPUNIT_TEST ( testSimpleIncomingIpCall ); CPPUNIT_TEST ( testSimpleIncomingIpCall );
// CPPUNIT_TEST ( testTwoOutgoingIpCall ); CPPUNIT_TEST ( testTwoOutgoingIpCall );
CPPUNIT_TEST ( testTwoIncomingIpCall ); // CPPUNIT_TEST ( testTwoIncomingIpCall );
CPPUNIT_TEST_SUITE_END(); CPPUNIT_TEST_SUITE_END();
public: public:
......
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